Audio source coding techniques can be divided into two classes: natural audio coding and speech coding. Natural audio coding is commonly used for music or arbitrary signals at medium bitrates, and generally offers wide audio bandwidth. Speech coders are basically limited to speech reproduction but can on the other hand be used at very low bitrates, albeit with low audio bandwidth. Wideband speech offers a major subjective quality improvement over narrow band speech. Increasing the bandwidth not only improves intelligibility and naturalness of speech, but also facilitates speaker recognition. Wideband speech coding is thus an important issue in next generation telephone systems. Further, due to the tremendous growth of the multimedia field, transmission of music and other non-speech signals at high quality over telephone systems is a desirable feature.
A high-fidelity linear PCM signal is very inefficient in terms of bitrate versus the perceptual entropy. The CD standard dictates 44.1 kHz sampling frequency, 16 bits per sample resolution and stereo. This equals a bitrate of 1411 kbit/s. To drastically reduce the bitrate, source coding can be performed using split-band perceptual audio codecs. These natural audio codecs exploit perceptual irrelevancy and statistical redundancy in the signal. Using the best codec technology, approximately 90% data reduction can be achieved for a standard CD-format signal with practically no perceptible degradation. Very high sound quality in stereo is thus possible at around 96 kbit/s, i.e. a compression factor of approximately 15:1. Some perceptual codecs offer even higher compression ratios. To achieve this, it is common to reduce the sample-rate and thus the audio bandwidth. It is also common to decrease the number of quantization levels, allowing occasionally audible quantization distortion, and to employ degradation of the stereo field, through intensity coding. Excessive use of such methods results in annoying perceptual degradation. Current codec technology is near saturation and further progress in coding gain is not expected. In order to improve the coding performance further, a new approach is necessary.
The human voice and most musical instruments generate quasistationary signals that emerge from oscillating systems. According to Fourier theory, any periodic signal may be expressed as a sum of sinusoids with the frequencies f, 2f, 3f, 4f, 5f etc. where f is the fundamental frequency. The frequencies form a harmonic series. A bandwidth limitation of such a signal is equivalent to a truncation of the harmonic series. Such a truncation alters the perceived timbre, tone colour, of a musical instrument or voice, and yields an audio signal that will sound “muffled” or “dull”, and intelligibility may be reduced. The high frequencies are thus important for the subjective impression of sound quality.
Prior art methods are mainly intended for improvement of speech codec performance and in particular intended for High Frequency Regeneration (HFR), an issue in speech coding. Such methods employ broadband linear frequency shifts, non-linearities or aliasing [U.S. Pat. No. 5,127,0541] generating intermodulation products or other non-harmonic frequency components which cause severe dissonance when applied to music signals. Such dissonance is referred to in the speech coding literature as “harsh” and “rough” sounding. Other synthetic speech HFR methods generate sinusoidal harmonics that are based on fundamental pitch estimation and are thus limited to tonal stationary sounds [U.S. Pat. No. 4,771,465]. Such prior art methods, although useful for low-quality speech applications, do not work for high quality speech or music signals. A few methods attempt to improve the performance of high quality audio source codecs. One uses synthetic noise signals generated at the decoder to substitute noise-like signals in speech or music previously discarded by the encoder [“Improving Audio Codecs by Noise Substitution” D. Schultz, JAES, Vol. 44, No. 7/8, 1996]. This is performed within an otherwise normally transmitted highband at an intermittent basis when noise signals are present. Another method recreates some missing highband harmonics that were lost in the coding process [“Audio Spectral Coder” A. J. S. Ferreira, AES Preprint 4201, 100th Convention, May 11-14 1996, Copenhagen] and is again dependent on tonal signals and pitch detection. Both methods operate at a low duty-cycle basis offering comparatively limited coding or performance gain.